A packet based network along with a centralized Call Server has been proposed as a means of providing system functionality such as Toll-Tandem switching within a conventional Public Telephone Network. The packet network may be based on different Protocols such as ATM, Frame Relay or IP. The architecture, which will be described in greater detail later with relation to FIG. 1 , consists basically of Interworking Functions (IWF), a Call Server, and a packet transport network. In this application these elements are configured to operate within the environment of an existing Public Switched Telephone Network (PSTN) consisting of Time Division Multiplexing (TDM) voice switches and a Signaling System #7 (SS7) network. Interworking Functions perform translation between conventional TDM signals and packets (e.g. ATM cells for VToA) at the interface between the TDM voice switches and the packet network. A centralized Call Server interacts with the SS7 signaling system translating Integrated Services Digital Network (ISDN) User Part (ISUP) messages concerning call setup into instructions to create connections (such as ATM Switched Virtual Circuits (SVCs)) between the Interworking Functions. For the end-to-end call setup, the Call Server generates the necessary SS7 signaling to the TDM voice switches normally provided by conventional Toll-Tandem switches. In its operation, the Call Server is also capable of performing many traditional Toll-Tandem functions such as number translation involving, for example 1-800 numbers.
It is possible that the existing TDM voice switches in such a system will be provided by a variety of vendors. Further, only a portion of these voice switches may fall under the administrative control of the network operator of the Voice over Packet (VoP) system.
A possible drawback of the above architecture, as in the conventional PSTN, is widespread network congestion under mass-calling situations. In this context a mass calling situation will exist when a particular number of geographic areas receive an excessive number of calls within a short time frame. While the dynamic nature of the routing architecture makes efficient use of transport resources and is highly adaptable, it makes application of conventional manual mass calling controls more difficult and if not addressed, mass calling has the potential to create network congestion at an accelerated rate.
There exists known prior art in the area of mass calling and congestion control for traditional TDM Voice Telephony. None of the prior art, of which applicants are aware, addresses the specific application area contemplated by the present invention, i.e. Mass Calling Control for Voice and Telephony over a Packet system wherein the network architecture employs a central Call Server.
U.S. Pat. 5,828,729 which issued Oct. 27, 1998 to Clermont et al, describes a method for detecting mass calling which operates by:
a) counting the number of call attempts towards a specific DN (Directory Number); and
b) counting the number of call release messages arising from unsuccessful call attempts to a DN.
The method of the '729 patent relies on monitoring the incoming and outgoing signaling links of the TDM switches and monitoring the Signaling Transfer Points (STP). A central controller then consolidates these measurements on a network level basis. The present invention, however is concerned with the use of a centralized Call Server which can handle call setup operations in its administrative region. The Call Server receives SS7 ISUP messages for inter-office calls originating, terminating, and tandeming through its administrative region. The architecture described herein performs detection at this point eliminating the need to place monitoring equipment on the signaling links of TDM switches (these are the Link Interface Devices (LID) referred to in FIG. 3c. of the aforementioned U.S. Pat. No. 5,828,729). The Call Server described herein also acts as a central point to consolidate information for the administrative region. The approach described herein also performs detection when service tones are handled at the destination rather than the originating switch. The above referenced US Patent describes detection but does not include an automatic response to the Mass Calling Event.
U.S. Pat. No. 5,923,742, which issued Jul. 13, 1999 to Kodialam et al, describes a method for detecting mass calling which operates by using two register and counter pairs and configurable thresholds. The mechanism described generates a count of the number of occurrences of a dialed number. The present invention involves a mechanism for consolidating network level information on failed call attempts. U.S. Pat. No. 5,923,742 describes a detection process but only responds by generating an alarm.
U.S. Pat. No. 5,295,183 which issued Mar. 15, 1994 to Langlois et al, describes enhancements to an earlier patent (U.S. Pat. No. 4,284,852 to Szybicki et al) concerning dynamic call routing. The restrictive control policy of the '183 patent, provides for a procedure known as call gapping. The congestion control policy operates in an environment where switching offices are connected to a central processor. These switching offices provide updates of congestion information to the central processor. For the restrictive control policy, the central processor, during an update cycle, identifies “restricted” destinations. It also determines a recommendation for staggering calls to the restricted destinations, i.e. call gapping.
The Call Server described herein detects failed call attempts at the SS7 protocol level. Unlike the prior art it does not rely on the ability of switching offices to provide congestion information. This is necessary to ensure that the VoP solution can operate in a multi-vendor environment of existing TDM voice switches. The Call Server participates in the call setup operations within its administrative region. As a result, it can carry out staggering or gapping operations itself without relying on the TDM voice switches. This may be necessary for traffic from a foreign network whose destination is a local subscriber undergoing mass calling in an environment where foreign network falls under a different administrative authority than the operator (e.g. possibly a competing carrier).
For regulation at the Call Server, a mechanism known herein as a credit bucket call approach, which will be described later, is simpler to implement than gapping as described in U.S. Pat. No. 5,295,183. Gapping requires a fetch of a timestamp on the arrival of a new call. The operation of the present invention achieves the same effect as gapping without the need to fetch a timestamp on the arrival of a new Initial Address Message (IAM). Instead only a periodic timer and a count are required. This bucket resembles a packet policer although the application domain is different.
U.S. Pat. No. 5,832,064, which issued Nov. 3, 1998 to Jeong, discloses a mass calling processing method for a televoting service. The '064 patent relies on an intelligent type service provider to offer a mass calling process in response to a known event of an influx of calls to a specific directory number. The method proposed in the present invention is intended for general telephony over a packet network.
U.S. Pat. No. 5,450,483, which issued Sep. 12, 1995 to Williams, is similar to the aforementioned U.S. Pat. No. 5,828,729 in that it also relies on counting failed calls to a called number. When the count exceeds a threshold, the counter is said to be in an overloaded state and call restriction is applied. After entering an overload state, if the counter falls below a second threshold then the counter goes into a no-overload state. The U.S. Pat. No. 5,450,483 patent uses a node level load detection and restriction in contrast to the centralized detection and restriction method described herein. The invention described herein includes a more resource efficient call restriction technique and operates in an environment of existing TDM voice switches from different vendors and does not require the existing voice switches to be modified in any manner. Detection based solely on a configured threshold of failed call attempts does not take into consideration that different destinations may have different numbers of voice lines. Since the destination may be outside the administrative region under control, the exact number of lines at the destination is unknown. The method described herein uses an approximation to the percent of call attempts failed as a mechanism for the Call Server to invoke regulation.
There also exist related techniques which cover some aspects of this problem. Specifically, code gapping is one technique used at Signaling Control Points (SCPs). In this approach excessive requests for translation services (e.g. 800 numbers) on a dialed number causes invocation of call regulation. Code gapping handles only mass calling to numbers related to translation services. The approach described herein handles both conventional numbers and numbers requiring translation services. This is a significant issue since a call to a Directory Number (DN) can arise not only from dialing an 800 number which translates into the DN but also from dialing the DN itself without invoking translation services (see U.S. Pat. No. 5,828,729 to Clermont et al.).
There exists known prior art in the area of Voice and Telephony over ATM. U.S. Pat. No. 5,568,475 which issued Oct. 22, 1996 to Doshi et. al, describes an ATM network architecture that interfaces with Synchronous Transfer Mode (STM) switches. Out of band signaling is used in this architecture to transfer telephone call signaling information. U.S. Pat. No. 5,568,475 does not specifically address the problem of Mass Calling Detection and Call Regulation. U.S. Pat. No. 5,956,334 which issued Sept. 21, 1999 to Chu et. al, presents an architecture for interfacing an ATM network to a telephony network. U.S. Pat. No. 5,956,334 also does not specifically address the problem of Mass Calling Detection and Call Regulation.